Cast: Javascript bindings for cast receiver

chrome/common/extensions/api/cast_streaming_receiver_session.idl

+// Copyright 2015 The Chromium Authors. All rights reserved.
+// Use of this source code is governed by a BSD-style license that can be
+// found in the LICENSE file.
+
+// The <code>chrome.cast.streaming.receiverSession</code> API creates a Cast
+// receiver session and adds the resulting audio and video tracks to a
+// MediaStream.
+namespace cast.streaming.receiverSession {
+  // The UDP socket address and port.
+  dictionary IPEndPoint {
+    DOMString address;
+    long port;
+  };
+
+  // Params for audio and video codec.
+  dictionary CodecSpecificParams {

[Yoyo Zhou, 2015/03/04 03:12:15]

Is this actually used?

[hubbe, 2015/03/04 23:12:33]

Not anymore -- deleted.

+    DOMString key;
+    DOMString value;
+  };
+
+  // RTP receiver parameters.
+  dictionary RtpReceiverParams {
+    // Maximum latency in milliseconds. This parameter controls the logic
+    // of flow control. Implementation can adjust latency adaptively and
+    // tries to keep it under this threshold. A larger value allows smoother
+    // playback at the cost of higher latency.
+    long maxLatency;
+
+    DOMString codecName;
+
+    // Synchronization source identifier.
+    long ssrc;
+
+    long feedbackSsrc;

[Yoyo Zhou, 2015/03/04 03:12:15]

Document this

[hubbe, 2015/03/04 23:12:33]

Done.

+
+    // Defaults to 48000 for audio and 90000 for video.

[Yoyo Zhou, 2015/03/04 03:12:15]

This should say something like the comment I found while trying to understand
this: "RTP timebase: The number of RTP units advanced per one second"

[hubbe, 2015/03/04 23:12:33]

Done.

+    long? rtpTimebase;
+
+    // 32 bytes hex-encoded AES key.
+    DOMString? aesKey;
+
+    // 32 bytes hex-encoded AES IV (Initialization vector) mask.
+    DOMString? aesIvMask;
+  };
+
+  callback ErrorCallback = void (DOMString error);
+
+  interface Functions {
+    // Creates a Cast receiver session which receives data from a UDP
+    // socket. The receiver will decode the incoming data into an audio
+    // and a video track which will be added to the provided media stream.
+    // The |audioParams| and |videoParams| is generally provided by the

[Yoyo Zhou, 2015/03/04 03:12:15]

nit: 'is' -> 'are'

[hubbe, 2015/03/04 23:12:33]

Done.

+    // sender through some other messaging channel.
+    //
+    // |audioParams| : the video stream parameters

[Yoyo Zhou, 2015/03/04 03:12:15]

The audio stream parameters.

[hubbe, 2015/03/04 23:12:33]

Done.

+    // |videoParams| : the video stream parameters

[Yoyo Zhou, 2015/03/04 03:12:15]

nit: Be consistent capitalizing and punctuating.

[hubbe, 2015/03/04 23:12:33]

Done.

+    // |localEndpoint| : Set this to bind() to a local UDP port.
+    // |height| : Video height
+    // |width| : Video width
+    // |maxFrameRate| : Max video frame rate.
+    // |mediaStreamURL| : URL of MediaStream to add the audio & video to.

[Yoyo Zhou, 2015/03/04 03:12:15]

nit: 'and' rather than '&'

[hubbe, 2015/03/04 23:12:33]

Done.

+    // |transport_options| : Optional transport settings.
+    [nocompile] static void createAndBind(
+        RtpReceiverParams audioParams,
+        RtpReceiverParams videoParams,
+        IPEndPoint localEndpoint,
+        long maxWidth,
+        long maxHeight,
+        double maxFrameRate,
+        DOMString mediaStreamURL,
+        ErrorCallback error_callback,
+        optional object transport_options);
+  };
+};